/*
    ALmixer:  A library to make playing pre-loaded sounds and streams easier
	with high performance and potential access to OpenAL effects.
    Copyright 2002, 2010 Eric Wing <ewing . public @ playcontrol.net>

    This library is free software; you can redistribute it and/or
    modify it under the terms of the GNU Library General Public
    License as published by the Free Software Foundation; either
    version 2 of the License, or (at your option) any later version.

    This library is distributed in the hope that it will be useful,
    but WITHOUT ANY WARRANTY; without even the implied warranty of
    MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
    Library General Public License for more details.

    You should have received a copy of the GNU Library General Public
    License along with this library; if not, write to the Free
    Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA

*/


 /**
 * @mainpage
 * ALmixer (which I sometimes call "SDL-OpenAL-Mixer" or "SDL_ALmixer") is a cross-platform audio library built 
 * on top of OpenAL to make playing and managing sounds easier. 
 * ALmixer provides a simple API inspired by SDL_mixer to make playing sounds easy 
 * with having to worry about directly dealing with OpenAL sources, buffers, 
 * and buffer queuing directly.
 * ALmixer currently utilizes SDL_sound behind the scenes to decode 
 * various audio formats such as WAV, MP3, AAC, MP4, OGG, etc.
 *
 * This library is targeted towards two major groups:
 * - People who just want an easy, high performance, way to play audio (don't care if its OpenAL or not)
 * - People who want to an easy way to play audio in OpenAL but still want access to OpenAL directly.
 *  
 * ALmixer exposes OpenAL sources in the API so you can freely use ALmixer 
 * in larger OpenAL applications that need to apply OpenAL 3D effects and features 
 * to playing sounds.
 *
 * The API is heavily influenced and inspired by SDL_mixer, though there is one major
 * conceptual design difference. ALmixer doesn't divide sound and music playback into two
 * separate play APIs. Instead, there is one unified play API and you specify via the 
 * load API whether you want the audio resource loaded as a stream or completely preloaded.
 * This allows you to have any arbitrary number of streaming sources playing simultaneously
 * (such as music and speech) unlike SDL_mixer where you are limited to only one "music" 
 * channel.
 *
 * A less major conceptual design difference is every "Channel" API has a corresponding "Source" API.
 * Every "channel" (in the SDL_mixer definition context) maps to a corresponding OpenAL source id. You can use
 * this source ID directly with OpenAL API commands to utilize OpenAL effects such as position, Doppler, etc.
 * Convenience APIs are provided to let you convert channel numbers to source ids and vice-versa.
 *
 * Another change which is a pet-peev of mine with SDL_mixer is the lack of a user_data parameter in callbacks.
 * ALmixer callbacks allow you to pass user_data (aka context) pointers through the callback functions.
 *
 * @note There are some #defines you can set to change the behavior at compile time. Most you shouldn't touch.
 * The one worth noting is ENABLE_ALMIXER_THREADS. If enabled, ALmixer_Update() is automatically called on a 
 * background thread so you no longer have to explicitly call it. (The function turns into a no-op so your existing
 * code won't break.) Having Update run in a separate thread has some advantages, particularly for streaming
 * audio as all the OpenAL buffer queuing happens in this function. It is less likely the background thread will
 * be blocked for long periods and thus less likely your buffer queues will be starved. However, this means you 
 * need to be extra careful about what you do in callback functions as they are invoked from the background thread.
 * I still consider this feature a experimental (though I am starting to use it more myself) and there
 * may still be bugs.
 *
 * @author Eric Wing
 *
 * Home Page: http://playcontrol.net/opensource/ALmixer
 */

/**
 * @file
 * ALmixer (which I sometimes call "SDL-OpenAL-Mixer" or "SDL_ALmixer") is a cross-platform audio library built 
 * on top of OpenAL to make playing and managing sounds easier. 
 * ALmixer provides a simple API inspired by SDL_mixer to make playing sounds easy 
 * with having to worry about directly dealing with OpenAL sources, buffers, 
 * and buffer queuing directly.
 * ALmixer currently utilizes SDL_sound behind the scenes to decode 
 * various audio formats such as WAV, MP3, AAC, MP4, OGG, etc.
 *
 * This library is targeted towards two major groups:
 * - People who just want an easy, high performance, way to play audio (don't care if its OpenAL or not)
 * - People who want to an easy way to play audio in OpenAL but still want access to OpenAL directly.
 *  
 * ALmixer exposes OpenAL sources in the API so you can freely use ALmixer 
 * in larger OpenAL applications that need to apply OpenAL 3D effects and features 
 * to playing sounds.
 *
 * The API is heavily influenced and inspired by SDL_mixer, though there is one major
 * conceptual design difference. ALmixer doesn't divide sound and music playback into two
 * separate play APIs. Instead, there is one unified play API and you specify via the 
 * load API whether you want the audio resource loaded as a stream or completely preloaded.
 * This allows you to have any arbitrary number of streaming sources playing simultaneously
 * (such as music and speech) unlike SDL_mixer where you are limited to only one "music" 
 * channel.
 *
 * A less major conceptual design difference is every "Channel" API has a corresponding "Source" API.
 * Every "channel" (in the SDL_mixer definition context) maps to a corresponding OpenAL source id. You can use
 * this source ID directly with OpenAL API commands to utilize OpenAL effects such as position, Doppler, etc.
 * Convenience APIs are provided to let you convert channel numbers to source ids and vice-versa.
 *
 * Another change which is a pet-peev of mine with SDL_mixer is the lack of a user_data parameter in callbacks.
 * ALmixer callbacks allow you to pass user_data (aka context) pointers through the callback functions.
 *
 * @note There are some #defines you can set to change the behavior at compile time. Most you shouldn't touch.
 * The one worth noting is ENABLE_ALMIXER_THREADS. If enabled, ALmixer_Update() is automatically called on a 
 * background thread so you no longer have to explicitly call it. (The function turns into a no-op so your existing
 * code won't break.) Having Update run in a separate thread has some advantages, particularly for streaming
 * audio as all the OpenAL buffer queuing happens in this function. It is less likely the background thread will
 * be blocked for long periods and thus less likely your buffer queues will be starved. However, this means you 
 * need to be extra careful about what you do in callback functions as they are invoked from the background thread.
 * I still consider this feature a experimental (though I am starting to use it more myself) and there
 * may still be bugs.
 *
 * @author Eric Wing
 *
 * Home Page: http://playcontrol.net/opensource/ALmixer
 */


#ifndef _SDL_ALMIXER_H_
#define _SDL_ALMIXER_H_


#ifndef DOXYGEN_SHOULD_IGNORE_THIS
/** @cond DOXYGEN_SHOULD_IGNORE_THIS */

/* Note: For Doxygen to produce clean output, you should set the 
 * PREDEFINED option to remove ALMIXER_DECLSPEC, ALMIXER_CALL, and
 * the DOXYGEN_SHOULD_IGNORE_THIS blocks.
 * PREDEFINED = DOXYGEN_SHOULD_IGNORE_THIS=1 ALMIXER_DECLSPEC= ALMIXER_CALL=
 */

#ifdef ALMIXER_COMPILE_WITHOUT_SDL
	#if defined(_WIN32)
		#if defined(ALMIXER_BUILD_LIBRARY)
			#define ALMIXER_DECLSPEC __declspec(dllexport)
		#else
			#define ALMIXER_DECLSPEC __declspec(dllimport)
		#endif
	#else
		#if defined(ALMIXER_BUILD_LIBRARY)
			#if defined (__GNUC__) && __GNUC__ >= 4
				#define ALMIXER_DECLSPEC __attribute__((visibility("default")))
			#else
				#define ALMIXER_DECLSPEC
			#endif
		#else
			#define ALMIXER_DECLSPEC
		#endif
	#endif

	#if defined(_WIN32)
		#define ALMIXER_CALL __cdecl
	#else
		#define ALMIXER_CALL
	#endif
#else
	#include "SDL_types.h" /* will include begin_code.h which is what I really want */
	#define ALMIXER_DECLSPEC DECLSPEC
	#define ALMIXER_CALL SDLCALL
#endif

/** @endcond DOXYGEN_SHOULD_IGNORE_THIS */
#endif /* DOXYGEN_SHOULD_IGNORE_THIS */



/* Needed for OpenAL types since altypes.h was removed in 1.1 */
#ifdef ANDROID_NDK
	#include <AL/al.h>
#else
	#include "al.h"
#endif

/* Set up for C function definitions, even when using C++ */
#ifdef __cplusplus
extern "C" {
#endif

#ifdef ALMIXER_COMPILE_WITHOUT_SDL
	/**
	 * Struct that contains the version information of this library.
	 * This represents the library's version as three levels: major revision
	 * (increments with massive changes, additions, and enhancements),
	 * minor revision (increments with backwards-compatible changes to the
	 * major revision), and patchlevel (increments with fixes to the minor
	 * revision).
	 * @see ALMIXER_VERSION, ALmixer_GetLinkedVersion
	 */
	typedef struct ALmixer_version
	{
		ALubyte major;
		ALubyte minor;
		ALubyte patch;
	} ALmixer_version;
#else
	#include "SDL_version.h"
	#define ALmixer_version SDL_version
#endif

/* Printable format: "%d.%d.%d", MAJOR, MINOR, PATCHLEVEL
 */
#define ALMIXER_MAJOR_VERSION		0
#define ALMIXER_MINOR_VERSION		1
#define ALMIXER_PATCHLEVEL			0


/** 
 * @defgroup CoreOperation Initialization, Tear-down, and Core Operational Commands
 * @{
 * Functions for setting up and using ALmixer.
 */
 
 
/**
 * This macro fills in a version structure with the version of the
 * library you compiled against. This is determined by what header the
 * compiler uses. Note that if you dynamically linked the library, you might
 * have a slightly newer or older version at runtime. That version can be
 * determined with ALmixer_GetLinkedVersion(), which, unlike 
 * ALMIXER_GET_COMPILED_VERSION, is not a macro.
 *
 * @note When compiled with SDL, this macro can be used to fill a version structure 
 * compatible with SDL_version.
 *
 * @param X A pointer to a ALmixer_version struct to initialize.
 *
 * @see ALmixer_version, ALmixer_GetLinkedVersion
 */
#define ALMIXER_GET_COMPILED_VERSION(X)                                           \
	{                                                                       \
		(X)->major = ALMIXER_MAJOR_VERSION;                          \
		(X)->minor = ALMIXER_MINOR_VERSION;                          \
		(X)->patch = ALMIXER_PATCHLEVEL;                             \
	}

/**
 * Gets the library version of the dynamically linked ALmixer you are using.
 * This gets the version of ALmixer that is linked against your program.
 * If you are using a shared library (DLL) version of ALmixer, then it is
 * possible that it will be different than the version you compiled against.
 *
 * This is a real function; the macro ALMIXER_GET_COMPILED_VERSION 
 * tells you what version of tErrorLib you compiled against:
 *
 * @code
 * ALmixer_version compiled;
 * ALmixer_version linked;
 *
 * ALMIXER_GET_COMPILED_VERSION(&compiled);
 * ALmixer_GetLinkedVersion(&linked);
 * printf("We compiled against tError version %d.%d.%d ...\n",
 *           compiled.major, compiled.minor, compiled.patch);
 * printf("But we linked against tError version %d.%d.%d.\n",
 *           linked.major, linked.minor, linked.patch);
 * @endcode
 *
 * @see ALmixer_version, ALMIXER_GET_COMPILED_VERSION
 */
extern ALMIXER_DECLSPEC const ALmixer_version* ALMIXER_CALL ALmixer_GetLinkedVersion(void);

#ifdef ALMIXER_COMPILE_WITHOUT_SDL
	/**
	 * Gets the last error string that was set by the system and clears the error.
	 *
	 * @note When compiled with SDL, this directly uses SDL_GetError.
	 * 
	 * @return Returns a string containing the last error or "" when no error is set.
	 */
	extern ALMIXER_DECLSPEC const char* ALMIXER_CALL ALmixer_GetError(void);
	/**
	 * Sets an error string that can be retrieved by ALmixer_GetError.
	 *
	 * @note When compiled with SDL, this directly uses SDL_SetError.
	 * 
	 * param The error string to set.
	 */
	extern ALMIXER_DECLSPEC void ALMIXER_CALL ALmixer_SetError(const char *fmt, ...);
#else
	#include "SDL_error.h"
	/**
	 * Gets the last error string that was set by the system and clears the error.
	 *
	 * @note When compiled with SDL, this directly uses SDL_GetError.
	 * 
	 * @return Returns a string containing the last error or "" when no error is set.
	 */
	#define ALmixer_GetError 	SDL_GetError
	/**
	 * Sets an error string that can be retrieved by ALmixer_GetError.
	 *
	 * @note When compiled with SDL, this directly uses SDL_SetError.
	 * 
	 * param The error string to set.
	 */
	#define ALmixer_SetError 	SDL_SetError
#endif


#ifdef ALMIXER_COMPILE_WITHOUT_SDL
	#include "ALmixer_rwops.h"
#else
	#include "SDL_rwops.h"
	/**
	 * A struct that mimicks the SDL_RWops structure.
	 *
	 * @note When compiled with SDL, this directly uses SDL_RWops.
	 */
	#define ALmixer_RWops 	SDL_RWops
#endif


#define ALMIXER_DEFAULT_FREQUENCY 	0
#define ALMIXER_DEFAULT_REFRESH 	0
#define ALMIXER_DEFAULT_NUM_CHANNELS	16
#define ALMIXER_DEFAULT_NUM_SOURCES		ALMIXER_DEFAULT_NUM_CHANNELS

/** 
 * This is the recommended Init function. This will initialize the context, SDL_sound,
 * and the mixer system. You should call this in the setup of your code, after SDL_Init.
 * If you attempt to bypass this function, you do so at your own risk.
 *
 * @note ALmixer expects the SDL audio subsystem to be disabled. In some cases, an enabled
 * SDL audio subsystem will interfere and cause problems in your app. This Init method explicitly
 * disables the SDL subsystem if SDL is compiled in. 
 *
 * @note The maximum number of sources is OpenAL implementation dependent.
 * Currently 16 is lowest common denominator for all OpenAL implementations in current use.
 * 32 is currently the second lowest common denominator.
 * If you try to allocate more sources than are actually available, this function may return false depending
 * if the OpenAL implementation returns an error or not. It is possible for OpenAL to silently fail
 * so be very careful about picking too many sources.
 *
 * @param playback_frequency The sample rate you want OpenAL to play at, e.g. 44100
 * Note that OpenAL is not required to actually respect this value.
 * Pass in 0 or ALMIXER_DEFAULT_FREQUENCY to specify you want to use your implementation's default value.
 * @param num_sources The number of OpenAL sources (also can be thought of as 
 * SDL_mixer channels) you wish to allocate.
 * Pass in 0 or ALMIXER_DEFAULT_NUM_SOURCES to use ALmixer's default value.
 * @param refresh_rate The refresh rate you want OpenAL to operate at. 
 * Note that OpenAL is not required to respect this value.
 * Pass in 0 or ALMIXER_DEFAULT_REFRESH to use OpenAL default behaviors.
 * @return Returns AL_FALSE on a failure or AL_TRUE if successfully initialized.
 */
extern ALMIXER_DECLSPEC ALboolean ALMIXER_CALL ALmixer_Init(ALuint playback_frequency, ALint num_sources, ALuint refresh_rate);

/** 
 * InitContext will only initialize the OpenAL context (and not the mixer part).
 * Note that SDL_Sound is also initialized here because load order matters
 * because SDL audio will conflict with OpenAL when using SMPEG. This is only 
 * provided as a backdoor and is not recommended.
 *
 * @note This is a backdoor in case you need to initialize the AL context and 
 * the mixer system separately. I strongly recommend avoiding these two functions
 * and use the normal Init() function.
 */
extern ALMIXER_DECLSPEC ALboolean ALMIXER_CALL ALmixer_InitContext(ALuint playback_frequency, ALuint refresh_rate);

/** 
 * InitMixer will only initialize the Mixer system. This is provided in the case 
 * that you need control over the loading of the context. You may load the context 
 * yourself, and then call this function. This is not recommended practice, but is 
 * provided as a backdoor in case you have good reason to 
 * do this. Be warned that if ALmixer_InitMixer() fails,
 * it will not clean up the AL context. Also be warned that Quit() still does try to 
 * clean up everything.
 *
 * @note This is a backdoor in case you need to initialize the AL context and 
 * the mixer system separately. I strongly recommend avoiding these two functions
 * and use the normal Init() function.
 */
extern ALMIXER_DECLSPEC ALboolean ALMIXER_CALL ALmixer_InitMixer(ALint num_sources);

/**
 * This shuts down ALmixer. Please remember to free your ALmixer_Data* instances
 * before calling this method.
 */
extern ALMIXER_DECLSPEC void ALMIXER_CALL ALmixer_Quit(void);
/**
 * Returns whether ALmixer has been initializatized (via Init) or not.
 * @return Returns true for initialized and false for not initialized.
 */
extern ALMIXER_DECLSPEC ALboolean ALMIXER_CALL ALmixer_IsInitialized(void);

/**
 * Returns the frequency that OpenAL is set to.
 * @note This function is not guaranteed to give correct information and is OpenAL implementation dependent.
 * @return Returns the frequency, e.g. 44100.
 */
extern ALMIXER_DECLSPEC ALuint ALMIXER_CALL ALmixer_GetFrequency(void);

/**
 * Let's you change the maximum number of channels/sources available.
 * This function is not heavily tested. It is probably better to simply initialize
 * ALmixer with the number of sources you want when you initialize it instead of 
 * dynamically changing it later.
 */
extern ALMIXER_DECLSPEC ALint ALMIXER_CALL ALmixer_AllocateChannels(ALint num_chans);

/**
 * Allows you to reserve a certain number of channels so they won't be automatically
 * allocated to play on.
 * This function will effectively block off a certain number of channels so they won't
 * be automatically assigned to be played on when you call various play functions
 * (applies to both play-channel and play-source functions since they are the same under the hood).
 * The lowest number channels will always be blocked off first.
 * For example, if there are 16 channels available, and you pass 2 into this function,
 * channels 0 and 1 will be reserved so they won't be played on automatically when you specify
 * you want to play a sound on any available channel/source. You can 
 * still play on channels 0 and 1 if you explicitly designate you want to play on their channel
 * number or source id.
 * Setting back to 0 will clear all the reserved channels so all will be available again for 
 * auto-assignment.
 * As an example, this feature can be useful if you always want your music to be on channel 0 and
 * speech on channel 1 and you don't want sound effects to ever occupy those channels. This allows
 * you to build in certain assumptions about your code, perhaps for deciding which data you want
 * to analyze in a data callback.
 * Specifying the number of reserve channels to the maximum number of channels will effectively
 * disable auto-assignment.
 * @param number_of_reserve_channels The number of channels/sources to reserve.
 * Or pass -1 to find out how many channels are currently reserved.
 * @return Returns the number of currently reserved channels.
 */
extern ALMIXER_DECLSPEC ALint ALMIXER_CALL ALmixer_ReserveChannels(ALint number_of_reserve_channels);


/**
 * The update function that allows ALmixer to update its internal state.
 * If not compiled with/using threads, this function must be periodically called
 * to poll ALmixer to force streamed music and other events to
 * take place. 
 * The typical place to put this function is in your main-loop. 
 * If threads are enabled, then this function just
 * returns 0 and is effectively a no-op. With threads, it is not necessary to call this function
 * because updates are handled internally on another thread. However, because threads are still considered
 * experimental, it is recommended you call this function in a proper place in your code in case
 * future versions of this library need to abandon threads.
 * @return Returns 0 if using threads. If not using threads, for debugging purposes, it returns
 * the number of buffers queued during the loop, or a negative value indicating the numer of errors encountered.
 * This is subject to change and should not be relied on.
 */
extern ALMIXER_DECLSPEC ALint ALMIXER_CALL ALmixer_Update(void);

/**
 * @}
 */
 
/** 
 * @defgroup LoadAPI Load Audio Functions
 * @{
 * Functions for loading and unloading audio data.
 */



/*
#define ALmixer_AudioInfo 	Sound_AudioInfo
*/

/*
#define ALMIXER_DEFAULT_BUFFERSIZE 32768
#define ALMIXER_DEFAULT_BUFFERSIZE 4096
*/
#define ALMIXER_DEFAULT_BUFFERSIZE 16384 

/* You probably never need to use these macros directly. */
#ifndef ALMIXER_DISABLE_PREDECODED_PRECOMPUTE_BUFFER_SIZE_OPTIMIZATION
	#define ALMIXER_DEFAULT_PREDECODED_BUFFERSIZE ALMIXER_DEFAULT_BUFFERSIZE * 4
#else
	/* I'm picking a smaller buffer because ALmixer will try to create a new larger buffer
	 * based on the length of the audio. So creating a large block up-front might just be a waste.
	 * However, if my attempts fail for some reason, this buffer size becomes a fallback.
	 * Having too small of a buffer might cause performance bottlenecks.
	 */
	#define ALMIXER_DEFAULT_PREDECODED_BUFFERSIZE 1024
#endif

/**
 * Specifies the maximum number of queue buffers to use for a sound stream.
 * Default Queue Buffers must be at least 2.
 */
#define ALMIXER_DEFAULT_QUEUE_BUFFERS 5
/**
 * Specifies the number of queue buffers initially filled when first loading a stream.
 * Default startup buffers should be at least 1. */
#define ALMIXER_DEFAULT_STARTUP_BUFFERS 2 

/*
#define ALMIXER_DECODE_STREAM 	0
#define ALMIXER_DECODE_ALL 		1
*/

/* This is a trick I picked up from Lua. Doing the typedef separately 
* (and I guess before the definition) instead of a single 
* entry: typedef struct {...} YourName; seems to allow me
* to use forward declarations. Doing it the other way (like SDL)
* seems to prevent me from using forward declarions as I get conflicting
* definition errors. I don't really understand why though.
*/
typedef struct ALmixer_Data ALmixer_Data;
typedef struct ALmixer_AudioInfo ALmixer_AudioInfo;

/**
 * Roughly the equvialent to the Sound_AudioInfo struct in SDL_sound.
 * Types have been changed to use AL types because I know those are available.
 * This is different than SDL which uses fixed types so there might be subtle
 * things you need to pay attention to..
 * @note Originally, I just used the Sound_AudioInfo directly, but
 * I've been trying to reduce the header dependencies for this file.
 * But more to the point, I've been interested in dealing with the 
 * WinMain override problem Josh faced when trying to use SDL components
 * in an MFC app which didn't like losing control of WinMain. 
 * My theory is that if I can purge the header of any thing that 
 * #include's SDL_main.h, then this might work.
 * So I am now introducing my own AudioInfo struct.
 */
struct ALmixer_AudioInfo
{
	ALushort format;  /**< Equivalent of SDL_AudioSpec.format. */
	ALubyte channels; /**< Number of sound channels. 1 == mono, 2 == stereo. */
	ALuint rate;    /**< Sample rate; frequency of sample points per second. */
};



/**
 * This is a general loader function to load an audio resource from an RWops.
 * Generally, you should use the LoadStream and LoadAll specializations of this function instead which call this.
 * @param rw_ops The rwops pointing to the audio resource you want to load.
 * @param file_ext The file extension of your audio type which is used as a hint by the backend to decide which
 * decoder to use.
 * @param buffer_size The size of a buffer to allocate for read chunks. This number should be in quantized with 
 * the valid frame sizes of your audio data. If the data is streamed, the data will be read in buffer_size chunks.
 * If the file is to be predecoded, optimizations may occur and this value might be ignored.
 * @param decode_mode_is_predecoded Specifies whether you want to completely preload the data or stream the data in chunks.
 * @param max_queue_buffers For streamed data, specifies the maximum number of buffers that can be queued at any given time.
 * @param num_startup_buffers For streamed data, specifies the number of buffers to fill before playback starts.
 * @param access_data A boolean that specifies if you want the data contained in the currently playing buffer to be handed
 * to you in a callback function. Note that for predecoded data, you get back the entire buffer in one callback when the 
 * audio first starts playing. With streamed data, you get the data in buffer_size chunks. Callbacks are not guarnanteed
 * to be perfectly in-sync as this is a best-effort implementaiton. There are memory and performance considerations for 
 * using this feature, so if you don't need data callbacks, you should pass false to this function.
 * @return Returns an ALmixer_Data* of the loaded sample or NULL if failed.
 */
extern ALMIXER_DECLSPEC ALmixer_Data* ALMIXER_CALL ALmixer_LoadSample_RW(ALmixer_RWops* rw_ops, const char* file_ext, ALuint buffer_size, ALboolean decode_mode_is_predecoded, ALuint max_queue_buffers, ALuint num_startup_buffers, ALboolean access_data);

#ifdef DOXYGEN_ONLY
/**
 * This is the loader function to load an audio resource from an RWops as a stream.
 * @param rw_ops The rwops pointing to the audio resource you want to load.
 * @param file_ext The file extension of your audio type which is used as a hint by the backend to decide which
 * decoder to use.
 * @param buffer_size The size of a buffer to allocate for read chunks. This number should be in quantized with 
 * the valid frame sizes of your audio data. If the data is streamed, the data will be read in buffer_size chunks.
 * @param max_queue_buffers For streamed data, specifies the maximum number of buffers that can be queued at any given time.
 * @param num_startup_buffers For streamed data, specifies the number of buffers to fill before playback starts.
 * @param access_data A boolean that specifies if you want the data contained in the currently playing buffer to be handed
 * to you in a callback function. Note that for predecoded data, you get back the entire buffer in one callback when the 
 * audio first starts playing. With streamed data, you get the data in buffer_size chunks. Callbacks are not guarnanteed
 * to be perfectly in-sync as this is a best-effort implementaiton. There are memory and performance considerations for 
 * using this feature, so if you don't need data callbacks, you should pass false to this function.
 * @return Returns an ALmixer_Data* of the loaded sample or NULL if failed.
 */
ALmixer_Data* ALmixer_LoadStream_RW(ALmixer_RWops* rw_ops, const char* file_ext, ALuint buffer_size, ALuint max_queue_buffers, ALuint num_startup_buffers, ALboolean access_data);
#else
#define ALmixer_LoadStream_RW(rw_ops, file_ext, buffer_size, max_queue_buffers, num_startup_buffers, access_data) ALmixer_LoadSample_RW(rw_ops,file_ext, buffer_size, AL_FALSE, max_queue_buffers, num_startup_buffers, access_data)
#endif

#ifdef DOXYGEN_ONLY
/**
 * This is the loader function to completely preload an audio resource from an RWops into RAM.
 * @param rw_ops The rwops pointing to the audio resource you want to load.
 * @param file_ext The file extension of your audio type which is used as a hint by the backend to decide which
 * decoder to use.
 * @param access_data A boolean that specifies if you want the data contained in the currently playing buffer to be handed
 * to you in a callback function. Note that for predecoded data, you get back the entire buffer in one callback when the 
 * audio first starts playing. With streamed data, you get the data in buffer_size chunks. Callbacks are not guarnanteed
 * to be perfectly in-sync as this is a best-effort implementaiton. There are memory and performance considerations for 
 * using this feature, so if you don't need data callbacks, you should pass false to this function.
 * @return Returns an ALmixer_Data* of the loaded sample or NULL if failed.
 */
ALmixer_Data* ALmixer_LoadAll_RW(ALmixer_RWops* rw_ops, const char* file_ext, ALboolean access_data);
#else
#define ALmixer_LoadAll_RW(rw_ops, file_ext, access_data) ALmixer_LoadSample_RW(rw_ops, fileext, ALMIXER_DEFAULT_PREDECODED_BUFFERSIZE, AL_TRUE, 0, 0, access_data)
#endif

/**
 * This is a general loader function to load an audio resource from a file.
 * Generally, you should use the LoadStream and LoadAll specializations of this function instead which call this.
 * @param file_name The file of the audio resource you want to load.
 * @param buffer_size The size of a buffer to allocate for read chunks. This number should be in quantized with 
 * the valid frame sizes of your audio data. If the data is streamed, the data will be read in buffer_size chunks.
 * If the file is to be predecoded, optimizations may occur and this value might be ignored.
 * @param decode_mode_is_predecoded Specifies whether you want to completely preload the data or stream the data in chunks.
 * @param max_queue_buffers For streamed data, specifies the maximum number of buffers that can be queued at any given time.
 * @param num_startup_buffers For streamed data, specifies the number of buffers to fill before playback starts.
 * @param access_data A boolean that specifies if you want the data contained in the currently playing buffer to be handed
 * to you in a callback function. Note that for predecoded data, you get back the entire buffer in one callback when the 
 * audio first starts playing. With streamed data, you get the data in buffer_size chunks. Callbacks are not guarnanteed
 * to be perfectly in-sync as this is a best-effort implementaiton. There are memory and performance considerations for 
 * using this feature, so if you don't need data callbacks, you should pass false to this function.
 * @return Returns an ALmixer_Data* of the loaded sample or NULL if failed.
 */
extern ALMIXER_DECLSPEC ALmixer_Data * ALMIXER_CALL ALmixer_LoadSample(const char* file_name, ALuint buffer_size, ALboolean decode_mode_is_predecoded, ALuint max_queue_buffers, ALuint num_startup_buffers, ALboolean access_data);

#ifdef DOXYGEN_ONLY
/**
 * This is the loader function to load an audio resource from a file.
 * @param file_name The file to the audio resource you want to load.
 * @param buffer_size The size of a buffer to allocate for read chunks. This number should be in quantized with 
 * the valid frame sizes of your audio data. If the data is streamed, the data will be read in buffer_size chunks.
 * @param max_queue_buffers For streamed data, specifies the maximum number of buffers that can be queued at any given time.
 * @param num_startup_buffers For streamed data, specifies the number of buffers to fill before playback starts.
 * @param access_data A boolean that specifies if you want the data contained in the currently playing buffer to be handed
 * to you in a callback function. Note that for predecoded data, you get back the entire buffer in one callback when the 
 * audio first starts playing. With streamed data, you get the data in buffer_size chunks. Callbacks are not guarnanteed
 * to be perfectly in-sync as this is a best-effort implementaiton. There are memory and performance considerations for 
 * using this feature, so if you don't need data callbacks, you should pass false to this function.
 * @return Returns an ALmixer_Data* of the loaded sample or NULL if failed.
 */
ALmixer_Data* ALmixer_LoadStream(const char* file_name, ALuint buffer_size, ALuint max_queue_buffers, ALuint num_startup_buffers, ALboolean access_data);
#else
#define ALmixer_LoadStream(file_name, buffer_size, max_queue_buffers, num_startup_buffers,access_data) ALmixer_LoadSample(file_name, buffer_size, AL_FALSE, max_queue_buffers, num_startup_buffers, access_data)
#endif

#ifdef DOXYGEN_ONLY
/**
 * This is the loader function to completely preload an audio resource from a file into RAM.
 * @param file_name The file to the audio resource you want to load.
 * @param access_data A boolean that specifies if you want the data contained in the currently playing buffer to be handed
 * to you in a callback function. Note that for predecoded data, you get back the entire buffer in one callback when the 
 * audio first starts playing. With streamed data, you get the data in buffer_size chunks. Callbacks are not guarnanteed
 * to be perfectly in-sync as this is a best-effort implementaiton. There are memory and performance considerations for 
 * using this feature, so if you don't need data callbacks, you should pass false to this function.
 * @return Returns an ALmixer_Data* of the loaded sample or NULL if failed.
 */
ALmixer_Data* ALmixer_LoadAll(const char* file_name, ALboolean access_data);
#else
#define ALmixer_LoadAll(file_name, access_data) ALmixer_LoadSample(file_name, ALMIXER_DEFAULT_PREDECODED_BUFFERSIZE, AL_TRUE, 0, 0, access_data)
#endif

/**
 * This is a back door general loader function for RAW samples or if you need to specify the ALmixer_AudioInfo field.
 * Use at your own risk.
 * Generally, you should use the LoadStream and LoadAll specializations of this function instead which call this.
 * @param rw_ops The rwops pointing to the audio resource you want to load.
 * @param file_ext The file extension of your audio type which is used as a hint by the backend to decide which
 * decoder to use. Pass "raw" for raw formats.
 * @param desired_format The format you want audio decoded to. NULL will pick a default for you.
 * @param buffer_size The size of a buffer to allocate for read chunks. This number should be in quantized with 
 * the valid frame sizes of your audio data. If the data is streamed, the data will be read in buffer_size chunks.
 * If the file is to be predecoded, optimizations may occur and this value might be ignored.
 * @param decode_mode_is_predecoded Specifies whether you want to completely preload the data or stream the data in chunks.
 * @param max_queue_buffers For streamed data, specifies the maximum number of buffers that can be queued at any given time.
 * @param num_startup_buffers For streamed data, specifies the number of buffers to fill before playback starts.
 * @param access_data A boolean that specifies if you want the data contained in the currently playing buffer to be handed
 * to you in a callback function. Note that for predecoded data, you get back the entire buffer in one callback when the 
 * audio first starts playing. With streamed data, you get the data in buffer_size chunks. Callbacks are not guarnanteed
 * to be perfectly in-sync as this is a best-effort implementaiton. There are memory and performance considerations for 
 * using this feature, so if you don't need data callbacks, you should pass false to this function.
 * @return Returns an ALmixer_Data* of the loaded sample or NULL if failed.
 */
extern ALMIXER_DECLSPEC ALmixer_Data * ALMIXER_CALL ALmixer_LoadSample_RAW_RW(ALmixer_RWops* rw_ops, const char* file_ext, ALmixer_AudioInfo* desired_format, ALuint buffer_size, ALboolean decode_mode_is_predecoded, ALuint max_queue_buffers, ALuint num_startup_buffers, ALboolean access_data);

#ifdef DOXYGEN_ONLY
/**
 * This is a back door stream loader function for RAW samples or if you need to specify the ALmixer_AudioInfo field.
 * Use at your own risk.
 * @param rw_ops The rwops pointing to the audio resource you want to load.
 * @param file_ext The file extension of your audio type which is used as a hint by the backend to decide which
 * decoder to use. Pass "raw" for raw formats.
 * @param desired_format The format you want audio decoded to. NULL will pick a default for you.
 * @param buffer_size The size of a buffer to allocate for read chunks. This number should be in quantized with 
 * the valid frame sizes of your audio data. If the data is streamed, the data will be read in buffer_size chunks.
 * If the file is to be predecoded, optimizations may occur and this value might be ignored.
 * @param max_queue_buffers For streamed data, specifies the maximum number of buffers that can be queued at any given time.
 * @param num_startup_buffers For streamed data, specifies the number of buffers to fill before playback starts.
 * @param access_data A boolean that specifies if you want the data contained in the currently playing buffer to be handed
 * to you in a callback function. Note that for predecoded data, you get back the entire buffer in one callback when the 
 * audio first starts playing. With streamed data, you get the data in buffer_size chunks. Callbacks are not guarnanteed
 * to be perfectly in-sync as this is a best-effort implementaiton. There are memory and performance considerations for 
 * using this feature, so if you don't need data callbacks, you should pass false to this function.
 * @return Returns an ALmixer_Data* of the loaded sample or NULL if failed.
 */
ALmixer_Data* ALmixer_LoadStream_RAW_RW(ALmixer_RWops* rw_ops, const char* file_ext, ALmixer_AudioInfo* desired_format, ALuint buffer_size, ALuint max_queue_buffers, ALuint num_startup_buffers, ALboolean access_data);
#else
#define ALmixer_LoadStream_RAW_RW(rw_ops, file_ext, desired_format, buffer_size, max_queue_buffers, num_startup_buffers, access_data) ALmixer_LoadSample_RAW_RW(rw_ops, file_ext, desired_format, buffer_size, AL_FALSE, max_queue_buffers, num_startup_buffers, access_data)
#endif

#ifdef DOXYGEN_ONLY
/**
 * This is a back door loader function for complete preloading RAW samples into RAM or if you need to specify the ALmixer_AudioInfo field.
 * Use at your own risk.
 * @param rw_ops The rwops pointing to the audio resource you want to load.
 * @param file_ext The file extension of your audio type which is used as a hint by the backend to decide which
 * decoder to use. Pass "raw" for raw formats.
 * @param desired_format The format you want audio decoded to. NULL will pick a default for you.
 * @param access_data A boolean that specifies if you want the data contained in the currently playing buffer to be handed
 * to you in a callback function. Note that for predecoded data, you get back the entire buffer in one callback when the 
 * audio first starts playing. With streamed data, you get the data in buffer_size chunks. Callbacks are not guarnanteed
 * to be perfectly in-sync as this is a best-effort implementaiton. There are memory and performance considerations for 
 * using this feature, so if you don't need data callbacks, you should pass false to this function.
 * @return Returns an ALmixer_Data* of the loaded sample or NULL if failed.
 */
ALmixer_Data* ALmixer_LoadAll_RAW_RW(ALmixer_RWops* rw_ops, const char* file_ext, ALmixer_AudioInfo* desired_format, ALboolean access_data);
#else
#define ALmixer_LoadAll_RAW_RW(rw_ops, file_ext, desired_format, access_data) ALmixer_LoadSample_RAW_RW(rw_ops, file_ext, desired_format, ALMIXER_DEFAULT_PREDECODED_BUFFERSIZE, AL_TRUE, 0, 0, access_data)
#endif

/**
 * This is a back door general loader function for RAW samples or if you need to specify the ALmixer_AudioInfo field.
 * Use at your own risk.
 * Generally, you should use the LoadStream and LoadAll specializations of this function instead which call this.
 * @param file_name The file to the audio resource you want to load. Extension should be "raw" for raw formats.
 * @param desired_format The format you want audio decoded to. NULL will pick a default for you.
 * @param buffer_size The size of a buffer to allocate for read chunks. This number should be in quantized with 
 * the valid frame sizes of your audio data. If the data is streamed, the data will be read in buffer_size chunks.
 * If the file is to be predecoded, optimizations may occur and this value might be ignored.
 * @param decode_mode_is_predecoded Specifies whether you want to completely preload the data or stream the data in chunks.
 * @param max_queue_buffers For streamed data, specifies the maximum number of buffers that can be queued at any given time.
 * @param num_startup_buffers For streamed data, specifies the number of buffers to fill before playback starts.
 * @param access_data A boolean that specifies if you want the data contained in the currently playing buffer to be handed
 * to you in a callback function. Note that for predecoded data, you get back the entire buffer in one callback when the 
 * audio first starts playing. With streamed data, you get the data in buffer_size chunks. Callbacks are not guarnanteed
 * to be perfectly in-sync as this is a best-effort implementaiton. There are memory and performance considerations for 
 * using this feature, so if you don't need data callbacks, you should pass false to this function.
 * @return Returns an ALmixer_Data* of the loaded sample or NULL if failed.
 */
extern ALMIXER_DECLSPEC ALmixer_Data * ALMIXER_CALL ALmixer_LoadSample_RAW(const char* file_name, ALmixer_AudioInfo* desired_format, ALuint buffer_size, ALboolean decode_mode_is_predecoded, ALuint max_queue_buffers, ALuint num_startup_buffers, ALboolean access_data);

#ifdef DOXYGEN_ONLY
/**
 * This is a back door stream loader function for RAW samples or if you need to specify the ALmixer_AudioInfo field.
 * Use at your own risk.
 * @param file_name The file to the audio resource you want to load.Extension should be "raw" for raw formats.
 * @param desired_format The format you want audio decoded to. NULL will pick a default for you.
 * @param buffer_size The size of a buffer to allocate for read chunks. This number should be in quantized with 
 * the valid frame sizes of your audio data. If the data is streamed, the data will be read in buffer_size chunks.
 * If the file is to be predecoded, optimizations may occur and this value might be ignored.
 * @param max_queue_buffers For streamed data, specifies the maximum number of buffers that can be queued at any given time.
 * @param num_startup_buffers For streamed data, specifies the number of buffers to fill before playback starts.
 * @param access_data A boolean that specifies if you want the data contained in the currently playing buffer to be handed
 * to you in a callback function. Note that for predecoded data, you get back the entire buffer in one callback when the 
 * audio first starts playing. With streamed data, you get the data in buffer_size chunks. Callbacks are not guarnanteed
 * to be perfectly in-sync as this is a best-effort implementaiton. There are memory and performance considerations for 
 * using this feature, so if you don't need data callbacks, you should pass false to this function.
 * @return Returns an ALmixer_Data* of the loaded sample or NULL if failed.
 */
ALmixer_Data* ALmixer_LoadStream_RAW(const char* file_name, ALmixer_AudioInfo* desired_format, ALuint buffer_size, ALuint max_queue_buffers, ALuint num_startup_buffers, ALboolean access_data);
#else
#define ALmixer_LoadStream_RAW(file_name, desired_format, buffer_size, max_queue_buffers, num_startup_buffers, access_data) ALmixer_LoadSample_RAW(file_name, desired_format, buffer_size, AL_FALSE, max_queue_buffers, num_startup_buffers, access_data)
#endif

#ifdef DOXYGEN_ONLY
/**
 * This is a back door loader function for complete preloading RAW samples into RAM or if you need to specify the ALmixer_AudioInfo field.
 * Use at your own risk.
 * @param file_name The file to the audio resource you want to load. Extension should be "raw" for raw formats.
 * @param desired_format The format you want audio decoded to. NULL will pick a default for you.
 * @param access_data A boolean that specifies if you want the data contained in the currently playing buffer to be handed
 * to you in a callback function. Note that for predecoded data, you get back the entire buffer in one callback when the 
 * audio first starts playing. With streamed data, you get the data in buffer_size chunks. Callbacks are not guarnanteed
 * to be perfectly in-sync as this is a best-effort implementaiton. There are memory and performance considerations for 
 * using this feature, so if you don't need data callbacks, you should pass false to this function.
 * @return Returns an ALmixer_Data* of the loaded sample or NULL if failed.
 */
ALmixer_Data* ALmixer_LoadAll_RAW(const char* file_name, ALmixer_AudioInfo* desired_format, ALboolean access_data);
#else
#define ALmixer_LoadAll_RAW(file_name, desired_format, access_data) ALmixer_LoadSample_RAW(file_name, desired_format, ALMIXER_DEFAULT_PREDECODED_BUFFERSIZE, AL_TRUE, 0, 0, access_data)
#endif

/**
 * Frees an ALmixer_Data.
 * Releases the memory associated with a ALmixer_Data. Use this when you are done playing the audio sample
 * and wish to release the memory.
 * @warning Do not try releasing data that is currently in use (e.g. playing, paused).
 * @warning Make sure to free your data before calling ALmixer_Quit. Do not free data aftter ALmixer_Quit().
 * @param almixer_data The ALmixer_Data* you want to free.
 */
extern ALMIXER_DECLSPEC void ALMIXER_CALL ALmixer_FreeData(ALmixer_Data* almixer_data);


/**
 * Returns true if the almixer_data was completely loaded into memory or false if it was loaded
 * as a stream.
 * @param almixer_data The audio resource you want to know about.
 * @return AL_TRUE is predecoded, or AL_FALSE if streamed.
 */
extern ALMIXER_DECLSPEC ALboolean ALMIXER_CALL ALmixer_IsPredecoded(ALmixer_Data* almixer_data);

/**
 * @}
 */
 
/** 
 * @defgroup CallbackAPI Callbacks
 * @{
 * Functions for callbacks
 */

/**
 * Allows you to set a callback for when a sound has finished playing on a channel/source.
 * @param playback_finished_callback The function you want to be invoked when a sound finishes.
 * The callback function will pass you back the channel number which just finished playing,
 * the OpenAL source id associated with the channel, the ALmixer_Data* that was played,
 * a boolean telling you whether a sound finished playing because it ended normally or because
 * something interrupted the playback (such as the user calling ALmixer_Halt*), and the
 * user_data supplied as the second parameter to this function.
 * @param which_chan The ALmixer channel that the data is currently playing on.
 * @param al_source The OpenAL source that the data is currently playing on.
 * @param almixer_data The ALmixer_Data that was played.
 * @param finished_naturally AL_TRUE if the sound finished playing because it ended normally 
 * or AL_FALSE because something interrupted playback (such as the user calling ALmixer_Halt*).
 * @param user_data This will be passed back to you in the callback.
 *
 * @warning You should not call other ALmixer functions in this callback. 
 * Particularly in the case of when compiled with threads, recursive locking
 * will occur which will lead to deadlocks. Also be aware that particularly in the 
 * threaded case, the callbacks may (and currently do) occur on a background thread.
 * One typical thread safe strategy is to set flags or schedule events to occur on the
 * main thread.
 * One possible exception to the no-calling ALmixer functions rule is ALmixer_Free. ALmixer_Free
 * currently does not lock so it might okay to call this to free your data. However, this is not
 * tested and not the expected pattern to be used.
 */
extern ALMIXER_DECLSPEC void ALMIXER_CALL ALmixer_SetPlaybackFinishedCallback(void (*playback_finished_callback)(ALint which_channel, ALuint al_source, ALmixer_Data* almixer_data, ALboolean finished_naturally, void* user_data), void* user_data);

/**
 * Allows you to set a callback for getting audio data.
 * This is a callback function pointer that when set, will trigger a function
 * anytime there is new data loaded for a sample. The appropriate load 
 * parameter must be set in order for a sample to appear here.
 * Keep in mind the the current backend implementation must do an end run
 * around OpenAL because OpenAL lacks support for this kind of thing.
 * As such, buffers are copied at decode time, and there is no attempt to do
 * fine grained timing syncronization. You will be provided the entire buffer
 * that is decoded regardless of length. So if you predecoded the entire 
 * audio file, the entire data buffer will be provided in a single callback.
 * If you stream the data, you will be getting chunk sizes that are the same as
 * what you specified the decode size to be. Unfortunely, this means if you 
 * pick smaller buffers, you get finer detail at the expense/risk of buffer 
 * underruns. If you decode more data, you have to deal with the syncronization
 * issues if you want to display the data during playback in something like an
 * oscilloscope.
 *
 * @warning You should not call other ALmixer functions in this callback. 
 * Particularly in the case of when compiled with threads, recursive locking
 * will occur which will lead to deadlocks. Also be aware that particularly in the 
 * threaded case, the callbacks may (and currently do) occur on a background thread.
 * One typical thread safe strategy is to set flags or schedule events to occur on the
 * main thread.
 * 
 * @param playback_data_callback The function you want called back.
 * @param which_channel The ALmixer channel that the data is currently playing on.
 * @param al_source The OpenAL source that the data is currently playing on.
 * @param pcm_data This is a pointer to the data buffer containing ALmixer's 
 * version of the decoded data. Consider this data as read-only. In the 
 * non-threaded backend, this data will persist until potentially the next call
 * to Update(). Currently, data buffers are preallocated and not destroyed
 * until FreeData() is called (though this behavior is subject to change),
 * but the contents will change when the buffer needs to be reused for a 
 * future callback. The buffer reuse is tied to the amount of buffers that
 * may be queued.
 * But assuming I don't change this, this may allow for some optimization
 * so you can try referencing data from these buffers without worrying 
 * about crashing. (You still need to be aware that the data could be 
 * modified behind the scenes on an Update().)
 * The data type listed is an signed 8-bit format, but the real data may
 * not actually be this. ALbyte was chosen as a convenience. If you have 
 * a 16 bit format, you will want to cast the data and divide the num_bytes by 2.
 * Typically, data is either Sint16. This seems to be a 
 * convention audio people seem to follow though I'm not sure what the 
 * underlying reasons (if any) are for this. I suspect that there may be 
 * some nice alignment/conversion property if you need to cast between ALbyte
 * and ALubyte.
 * 
 * @param num_bytes This is the total length of the data buffer. It presumes
 * that this length is measured for ALbyte. So if you have Sint16 data, you
 * should divide num_bytes by two if you access the data as Sint16.
 * 
 * @param frequency The frequency the data was decoded at.
 *
 * @param num_channels_in_sample 1 for mono, 2 for stereo. Not to be confused with the ALmixer which_channel.
 *
 * @param bit_depth Bits per sample. This is expected to be 8 or 16. This 
 * number will tell you if you if you need to treat the data buffer as 
 * 16 bit or not.
 * 
 * @param is_unsigned 1 if the data is unsigned, 0 if signed. Using this
 * combined with bit_depth will tell you if you need to treat the data
 * as ALubyte, ALbyte, ALuint, or ALint.
 *
 * @param decode_mode_is_predecoded This is here to tell you if the data was totally 
 * predecoded or loaded as a stream. If predecoded, you will only get 
 * one data callback per playback instance. (This might also be true for 
 * looping the same sample...I don't remember how it was implemented. 
 * Maybe this should be fixed.)
 * 0 (ALMIXER_DECODE_STREAM) for streamed.
 * 1 (ALMIXER_DECODE_ALL) for predecoded.
 *
 * @param length_in_msec This returns the total length (time) of the data 
 * buffer in milliseconds. This could be computed yourself, but is provided
 * as a convenince.
 *
 * @param user_data The user data you pass in will be passed back to you in the callback. 
 */
extern ALMIXER_DECLSPEC void ALMIXER_CALL ALmixer_SetPlaybackDataCallback(void (*playback_data_callback)(ALint which_channel, ALuint al_source, ALbyte* pcm_data, ALuint num_bytes, ALuint frequency, ALubyte num_channels_in_sample, ALubyte bit_depth, ALboolean is_unsigned, ALboolean decode_mode_is_predecoded, ALuint length_in_msec, void* user_data), void* user_data);

/**
 * @}
 */
 
 /** 
 * @defgroup PlayAPI Functions useful for playback.
 * @{
 * These are core functions that are useful for controlling playback.
 * Also see the Volume functions for additional playback functions and Query functions for additional information.
 */

/**
 * Returns the total time in milliseconds of the audio resource.
 * Returns the total time in milliseconds of the audio resource.
 * If the total length cannot be determined, -1 will be returned.
 * @param almixer_data The audio sample you want to know the total time of.
 * @return The total time in milliseconds or -1 if some kind of failure.
 */
extern ALMIXER_DECLSPEC ALint ALMIXER_CALL ALmixer_GetTotalTime(ALmixer_Data* almixer_data);

/** 
 * This function will look up the OpenAL source id for the corresponding channel number.
 * @param which_channel The channel which you want to find the corresponding OpenAL source id for.
 * If -1 was specified, an available source for playback will be returned.
 * @return The OpenAL source id corresponding to the channel. 0 if you specified an illegal channel value.
 * Or 0 if you specified -1 and no sources were currently available.
 * @note ALmixer assumes your OpenAL implementation does not use 0 as a valid source ID. While the OpenAL spec
 * does not disallow 0 for valid source ids, as of now, there are no known OpenAL implementations in use that 
 * use 0 as a valid source id (partly due to problems this has caused developers in the past).
 */
extern ALMIXER_DECLSPEC ALuint ALMIXER_CALL ALmixer_GetSource(ALint which_channel);

/**
 * This function will look up the channel for the corresponding source.
 * @param al_source The source id you want to find the corresponding channel number for.
 * If 0 is supplied, it will try to return the first channel not in use. 
 * Returns -1 on error, or the channel.
 */
extern ALMIXER_DECLSPEC ALint ALMIXER_CALL ALmixer_GetChannel(ALuint al_source);

/**
 * Will look for a channel available for playback.
 * Given a start channel number, the search will increase to the highest channel until it finds one available.
 * @param start_channel The channel number you want to start looking at.
 * @return A channel available or -1 if none could be found.
 */
extern ALMIXER_DECLSPEC ALint ALMIXER_CALL ALmixer_FindFreeChannel(ALint start_channel);



/**
 * Play a sound on a channel with a time limit.
 * Plays a sound on a channel and will auto-stop after a specified number of milliseconds.
 * @param which_channel Allows you to specify the specific channel you want to play on. 
 * Channels range from 0 to the (Number of allocated channels - 1). If you specify -1, 
 * an available channel will be chosen automatically for you.
 * @note While paused, the auto-stop clock will also be paused. This makes it easy to always stop
 * a sample by a predesignated length without worrying about whether the user paused playback which would 
 * throw off your calculations.
 * @param almixer_data The audio resource you want to play.
 * @param number_of_loops The number of times to loop (repeat) playing the data. 
 * 0 means the data will play exactly once without repeat. -1 means infinitely loop.
 * @param number_of_milliseconds The number of milliseconds that should be played until the sample is auto-stopped.
 * -1 means don't auto-stop playing and let the sample finish playing normally (or if looping is set to infinite, 
 * the sample will never stop playing).
 * @return Returns the channel that was selected for playback or -1 if no channels were available.
 */
extern ALMIXER_DECLSPEC ALint ALMIXER_CALL ALmixer_PlayChannelTimed(ALint which_channel, ALmixer_Data* almixer_data, ALint number_of_loops, ALint number_of_milliseconds);

#ifdef DOXYGEN_ONLY
/**
 * The same as ALmixer_PlayChannelTimed, but the sound is played without time limits.
 * @see ALmixer_PlayChannelTimed.
 */ 
ALint ALmixer_PlayChannelTimed(ALint which_channel, ALmixer_Data* almixer_data, ALint number_of_loops);
#else
#define ALmixer_PlayChannel(channel,data,loops) ALmixer_PlayChannelTimed(channel,data,loops,-1)
#endif


/**
 * Play a sound on an OpenAL source with a time limit.
 * Plays a sound on an OpenAL source and will auto-stop after a specified number of milliseconds.
 * @param al_source Allows you to specify the OpenAL source you want to play on. 
 * If you specify 0, an available source will be chosen automatically for you.
 * @note Source values are not necessarily continguous and their values are implementation dependent.
 * Always use ALmixer functions to determine source values.
 * @note While paused, the auto-stop clock will also be paused. This makes it easy to always stop
 * a sample by a predesignated length without worrying about whether the user paused playback which would 
 * throw off your calculations.
 * @param almixer_data The audio resource you want to play.
 * @param number_of_loops The number of times to loop (repeat) playing the data. 
 * 0 means the data will play exactly once without repeat. -1 means infinitely loop.
 * @param number_of_milliseconds The number of milliseconds that should be played until the sample is auto-stopped.
 * -1 means don't auto-stop playing and let the sample finish playing normally (or if looping is set to infinite, 
 * the sample will never stop playing).
 * @return Returns the OpenAL source that was selected for playback or 0 if no sources were available.
 */
extern ALMIXER_DECLSPEC ALuint ALMIXER_CALL ALmixer_PlaySourceTimed(ALuint al_source, ALmixer_Data* almixer_data, ALint number_of_loops, ALint number_of_milliseconds);

#ifdef DOXYGEN_ONLY
/**
 * The same as ALmixer_PlaySourceTimed, but the sound is played without time limits.
 * @see ALmixer_PlaySourceTimed.
 */ 
ALint ALmixer_PlayChannelTimed(ALuint al_source, ALmixer_Data* almixer_data, ALint number_of_loops);
#else
#define ALmixer_PlaySource(al_source, almixer_data, number_of_loops) ALmixer_PlaySourceTimed(al_source, almixer_data, number_of_loops, -1)
#endif

/**
 * Stops playback on a channel.
 * Stops playback on a channel and clears the channel so it can be played on again.
 * @note Callbacks will still be invoked, but the finished_naturally flag will be set to AL_FALSE.
 * @param which_channel The channel to halt or -1 to halt all channels.
 * @return The actual number of channels halted on success or -1 on error.
 */
extern ALMIXER_DECLSPEC ALint ALMIXER_CALL ALmixer_HaltChannel(ALint which_channel);

/**
 * Stops playback on a channel.
 * Stops playback on a channel and clears the channel so it can be played on again.
 * @note Callbacks will still be invoked, but the finished_naturally flag will be set to AL_FALSE.
 * @param al_source The source to halt or 0 to halt all sources.
 * @return The actual number of sources halted on success or -1 on error.
 */
extern ALMIXER_DECLSPEC ALint ALMIXER_CALL ALmixer_HaltSource(ALuint al_source);

/**
 * Rewinds the sound to the beginning for a given data.
 * Rewinds the actual data, but the effect
 * may not be noticed until the currently buffered data is played.
 * @param almixer_data The data to rewind.
 * @returns true on success or false on error.
 */
extern ALMIXER_DECLSPEC ALboolean ALMIXER_CALL ALmixer_RewindData(ALmixer_Data* almixer_data);

/**
 * Rewinds the sound to the beginning that is playing on a specific channel.
 * If decoded all, rewind will instantly rewind it. Data is not 
 * affected, so it will start at the "Seek"'ed positiond.
 * Streamed data will rewind the actual data, but the effect
 * may not be noticed until the currently buffered data is played.
 * @param which_channel The channel to rewind or -1 to rewind all channels.
 * @return The actual number of channels rewound on success or -1 on error.
 */
extern ALMIXER_DECLSPEC ALint ALMIXER_CALL ALmixer_RewindChannel(ALint which_channel);
/**
 * Rewinds the sound to the beginning that is playing on a specific source.
 * If decoded all, rewind will instantly rewind it. Data is not 
 * affected, so it will start at the "Seek"'ed positiond.
 * Streamed data will rewind the actual data, but the effect
 * may not be noticed until the currently buffered data is played.
 * @param al_source The source to rewind or 0 to rewind all sources.
 * @return The actual number of sources rewound on success or -1 on error.
 */
extern ALMIXER_DECLSPEC ALint ALMIXER_CALL ALmixer_RewindSource(ALuint al_source);

/**
 * Seek the sound for a given data.
 * Seeks the actual data to the given millisecond. It
 * may not be noticed until the currently buffered data is played.
 * @param almixer_data The data to seek on.
 * @param msec_pos The time position to seek to in the audio in milliseconds.
 * @returns true on success or false on error.
 */
extern ALMIXER_DECLSPEC ALboolean ALMIXER_CALL ALmixer_SeekData(ALmixer_Data* almixer_data, ALuint msec_pos);

/**
 * Seeks the sound to the beginning that is playing on a specific channel.
 * If decoded all, seek will instantly seek it. Data is not 
 * affected, so it will start at the "Seek"'ed positiond.
 * Streamed data will seek the actual data, but the effect
 * may not be noticed until the currently buffered data is played.
 * @param which_channel The channel to seek or -1 to seek all channels.
 * @return The actual number of channels rewound on success or -1 on error.
 */
extern ALMIXER_DECLSPEC ALint ALMIXER_CALL ALmixer_SeekChannel(ALint which_channel, ALuint msec_pos);
/**
 * Seeks the sound to the beginning that is playing on a specific source.
 * If decoded all, seek will instantly seek it. Data is not 
 * affected, so it will start at the "Seek"'ed positiond.
 * Streamed data will seek the actual data, but the effect
 * may not be noticed until the currently buffered data is played.
 * @param al_source The source to seek or 0 to seek all sources.
 * @return The actual number of sources rewound on success or -1 on error.
 */
extern ALMIXER_DECLSPEC ALint ALMIXER_CALL ALmixer_SeekSource(ALuint al_source, ALuint msec_pos);

/**
 * Pauses playback on a channel.
 * Pauses playback on a channel. Should have no effect on channels that aren't playing.
 * @param which_channel The channel to pause or -1 to pause all channels.
 * @return The actual number of channels paused on success or -1 on error.
 */
extern ALMIXER_DECLSPEC ALint ALMIXER_CALL ALmixer_PauseChannel(ALint which_channel);
/**
 * Pauses playback on a source.
 * Pauses playback on a source. Should have no effect on source that aren't playing.
 * @param al_source The source to pause or -1 to pause all source.
 * @return The actual number of source paused on success or -1 on error.
 */
extern ALMIXER_DECLSPEC ALint ALMIXER_CALL ALmixer_PauseSource(ALuint al_source);

/**
 * Resumes playback on a channel that is paused.
 * Resumes playback on a channel that is paused. Should have no effect on channels that aren't paused.
 * @param which_channel The channel to resume or -1 to resume all channels.
 * @return The actual number of channels resumed on success or -1 on error.
 */
extern ALMIXER_DECLSPEC ALint ALMIXER_CALL ALmixer_ResumeChannel(ALint which_channel);

/**
 * Resumes playback on a source that is paused.
 * Resumes playback on a source that is paused. Should have no effect on sources that aren't paused.
 * @param al_source The source to resume or -1 to resume all sources.
 * @return The actual number of sources resumed on success or -1 on error.
 */
 extern ALMIXER_DECLSPEC ALint ALMIXER_CALL ALmixer_ResumeSource(ALuint al_source);

 
/**
 * Will cause a currently playing channel to stop playing in the specified number of milliseconds.
 * Will cause a currently playing channel to stop playing in the specified number of milliseconds.
 * This will override the value that was set when PlayChannelTimed or PlaySourceTimed was called
 * or override any previous calls to ExpireChannel or ExpireSource.
 * @param which_channel The channel to expire or -1 to apply to all channels.
 * @param number_of_milliseconds How many milliseconds from now until the expire triggers.
 * @return The actual number of channels this action is applied to on success or -1 on error.
 */
extern ALMIXER_DECLSPEC ALint ALMIXER_CALL ALmixer_ExpireChannel(ALint which_channel, ALint number_of_milliseconds);
/**
 * Will cause a currently playing source to stop playing in the specified number of milliseconds.
 * Will cause a currently playing source to stop playing in the specified number of milliseconds.
 * This will override the value that was set when PlayChannelTimed or PlaySourceTimed was called
 * or override any previous calls to ExpireChannel or ExpireSource.
 * @param al_source The source to expire or 0 to apply to all sources.
 * @param number_of_milliseconds How many milliseconds from now until the expire triggers.
 * @return The actual number of sources this action is applied to on success or -1 on error.
 */
extern ALMIXER_DECLSPEC ALint ALMIXER_CALL ALmixer_ExpireSource(ALuint al_source, ALint number_of_milliseconds);

/**
 * @}
 */

/** 
 * @defgroup VolumeAPI Volume and Fading
 * @{
 * Fade and volume functions directly call OpenAL functions related to AL_GAIN.
 * These functions are provided mostly for those who just want to play audio but are not planning
 * to use OpenAL features directly.
 * If you are using OpenAL directly (e.g. for 3D effects), you may want to be careful about using these as
 * they may fight/override values you directly set yourself.
 */

/**
 * Similar to ALmixer_PlayChannelTimed except that sound volume fades in from the minimum volume to the current AL_GAIN over the specified amount of time.
 * @see ALmixer_PlayChannelTimed.
 */
extern ALMIXER_DECLSPEC ALint ALMIXER_CALL ALmixer_FadeInChannelTimed(ALint which_channel, ALmixer_Data* almixer_data, ALint number_of_loops, ALuint fade_ticks, ALint expire_ticks);

#ifdef DOXYGEN_ONLY
/**
 * The same as ALmixer_FadeInChannelTimed, but the sound is played without time limits.
 * @see ALmixer_FadeInChannelTimed, ALmixer_PlayChannel.
 */ 
ALint ALmixer_FadeInChannel(ALint which_channel, ALmixer_Data* almixer_data, ALint number_of_loops, ALuint fade_ticks);
#else
#define ALmixer_FadeInChannel(which_channel, almixer_data, number_of_loops, fade_ticks) ALmixer_FadeInChannelTimed(which_channel, almixer_data, number_of_loops, fade_ticks, -1)
#endif

/**
 * Similar to ALmixer_PlaySourceTimed except that sound volume fades in from the minimum volume to the max volume over the specified amount of time.
 * @see ALmixer_PlaySourceTimed.
 */
extern ALMIXER_DECLSPEC ALuint ALMIXER_CALL ALmixer_FadeInSourceTimed(ALuint al_source, ALmixer_Data* almixer_data, ALint number_of_loops, ALuint fade_ticks, ALint expire_ticks);

#ifdef DOXYGEN_ONLY
/**
 * The same as ALmixer_FadeInSourceTimed, but the sound is played without time limits.
 * @see ALmixer_FadeInSourceTimed, ALmixer_PlaySource.
 */ 
extern ALuint ALmixer_FadeInSource(ALuint al_source, ALmixer_Data* almixer_data, ALint number_of_loops, ALuint fade_ticks);
#else
#define ALmixer_FadeInSource(al_source, almixer_data, number_of_loops, fade_ticks) ALmixer_FadeInSourceTimed(al_source, almixer_data, number_of_loops, fade_ticks, -1)
#endif

/**
 * Fade out a current playing channel.
 * Will fade out a currently playing channel over the specified period of time starting from now. 
 * The volume will be changed from the current AL_GAIN level to the AL_MIN_GAIN. 
 * The volume fade will interpolate over the specified period of time. 
 * The playback will halt at the end of the time period.
 * @param which_channel The channel to fade or -1 to fade all playing channels.
 * @param fade_ticks In milliseconds, the amount of time the fade out should take to complete.
 * @return Returns -1 on error or the number of channels faded.
 */
extern ALMIXER_DECLSPEC ALint ALMIXER_CALL ALmixer_FadeOutChannel(ALint which_channel, ALuint fade_ticks);

/**
 * Fade out a current playing source.
 * Will fade out a currently playing source over the specified period of time starting from now. 
 * The volume will be changed from the current AL_GAIN level to the AL_MIN_GAIN. 
 * The volume fade will interpolate over the specified period of time. 
 * The playback will halt at the end of the time period.
 * @param al_source The source to fade or -1 to fade all playing sources.
 * @param fade_ticks In milliseconds, the amount of time the fade out should take to complete.
 * @return Returns -1 on error or the number of sources faded.
 */
extern ALMIXER_DECLSPEC ALint ALMIXER_CALL ALmixer_FadeOutSource(ALuint al_source, ALuint fade_ticks);

/**
 * Gradually changes the volume from the current AL_GAIN to the specified volume.
 * Gradually changes the volume from the current AL_GAIN to the specified volume over the specified period of time.
 * This is some times referred to as volume ducking. 
 * Note that this function works for setting the volume higher as well as lower.
 * @param which_channel The channel to fade or -1 to fade all playing channels.
 * @param fade_ticks In milliseconds, the amount of time the volume change should take to complete.
 * @param volume The volume to change to. Valid values are 0.0 to 1.0.
 * @return Returns -1 on error or the number of channels faded.
 */
extern ALMIXER_DECLSPEC ALint ALMIXER_CALL ALmixer_FadeChannel(ALint which_channel, ALuint fade_ticks, ALfloat volume);

/**
 * Gradually changes the volume from the current AL_GAIN to the specified volume.
 * Gradually changes the volume from the current AL_GAIN to the specified volume over the specified period of time.
 * This is some times referred to as volume ducking. 
 * Note that this function works for setting the volume higher as well as lower.
 * @param al_source The source to fade or -1 to fade all playing sources.
 * @param fade_ticks In milliseconds, the amount of time the volume change should take to complete.
 * @param volume The volume to change to. Valid values are 0.0 to 1.0.
 * @return Returns -1 on error or the number of sources faded.
 */
extern ALMIXER_DECLSPEC ALint ALMIXER_CALL ALmixer_FadeSource(ALuint al_source, ALuint fade_ticks, ALfloat volume);

/**
 * Sets the volume via the AL_GAIN source property.
 * Sets the volume for a given channel via the AL_GAIN source property.
 * @param which_channel The channel to set the volume to or -1 to set on all channels.
 * @param volume The new volume to use. Valid values are 0.0 to 1.0.
 * @return AL_TRUE on success or AL_FALSE on error.
 */
extern ALMIXER_DECLSPEC ALboolean ALMIXER_CALL ALmixer_SetVolumeChannel(ALint which_channel, ALfloat volume);

/**
 * Sets the volume via the AL_GAIN source property.
 * Sets the volume for a given source via the AL_GAIN source property.
 * @param al_source The source to set the volume to or 0 to set on all sources.
 * @param volume The new volume to use. Valid values are 0.0 to 1.0.
 * @return AL_TRUE on success or AL_FALSE on error.
 */
extern ALMIXER_DECLSPEC ALboolean ALMIXER_CALL ALmixer_SetVolumeSource(ALuint al_source, ALfloat volume);

/**
 * Gets the volume via the AL_GAIN source property.
 * Gets the volume for a given channel via the AL_GAIN source property.
 * @param which_channel The channel to get the volume from.
 * -1 will return the average volume set across all channels.
 * @return Returns the volume for the specified channel, or the average set volume for all channels, or -1.0 on error.
 */
extern ALMIXER_DECLSPEC ALfloat ALMIXER_CALL ALmixer_GetVolumeChannel(ALint which_channel);

/**
 * Gets the volume via the AL_GAIN source property.
 * Gets the volume for a given source via the AL_GAIN source property.
 * @param al_source The source to get the volume from.
 * -1 will return the average volume set across all source.
 * @return Returns the volume for the specified source, or the average set volume for all sources, or -1.0 on error.
 */
extern ALMIXER_DECLSPEC ALfloat ALMIXER_CALL ALmixer_GetVolumeSource(ALuint al_source);

/**
 * Sets the maximum volume via the AL_MAX_GAIN source property.
 * Sets the maximum volume for a given channel via the AL_MAX_GAIN source property.
 * Max volumes will be clamped to this value.
 * @param which_channel The channel to set the volume to or -1 to set on all channels.
 * @param volume The new volume to use. Valid values are 0.0 to 1.0.
 * @return AL_TRUE on success or AL_FALSE on error.
 */
extern ALMIXER_DECLSPEC ALboolean ALMIXER_CALL ALmixer_SetMaxVolumeChannel(ALint which_channel, ALfloat volume);

/**
 * Sets the maximum volume via the AL_MAX_GAIN source property.
 * Sets the maximum volume for a given source via the AL_MAX_GAIN source property.
 * @param al_source The source to set the volume to or 0 to set on all sources.
 * @param volume The new volume to use. Valid values are 0.0 to 1.0.
 * @return AL_TRUE on success or AL_FALSE on error.
 */
extern ALMIXER_DECLSPEC ALboolean ALMIXER_CALL ALmixer_SetMaxVolumeSource(ALuint al_source, ALfloat volume);

/**
 * Gets the max volume via the AL_MAX_GAIN source property.
 * Gets the max volume for a given channel via the AL_MAX_GAIN source property.
 * @param which_channel The channel to get the volume from.
 * -1 will return the average volume set across all channels.
 * @return Returns the volume for the specified channel, or the average set volume for all channels, or -1.0 on error.
 */
extern ALMIXER_DECLSPEC ALfloat ALMIXER_CALL ALmixer_GetMaxVolumeChannel(ALint which_channel);

/**
 * Gets the maximum volume via the AL_MAX_GAIN source property.
 * Gets the maximum volume for a given source via the AL_MAX_GAIN source property.
 * @param al_source The source to set the volume to or 0 to set on all sources.
 * 0 will return the average volume set across all channels.
 * @return Returns the volume for the specified channel, or the average set volume for all channels, or -1.0 on error.
 */
extern ALMIXER_DECLSPEC ALfloat ALMIXER_CALL ALmixer_GetMaxVolumeSource(ALuint al_source);

/**
 * Sets the minimum volume via the AL_MIN_GAIN source property.
 * Sets the minimum volume for a given channel via the AL_MIN_GAIN source property.
 * Min volumes will be clamped to this value.
 * @param which_channel The channel to set the volume to or -1 to set on all channels.
 * @param volume The new volume to use. Valid values are 0.0 to 1.0.
 * @return AL_TRUE on success or AL_FALSE on error.
 */
extern ALMIXER_DECLSPEC ALboolean ALMIXER_CALL ALmixer_SetMinVolumeChannel(ALint which_channel, ALfloat volume);

/**
 * Sets the minimum volume via the AL_MIN_GAIN source property.
 * Sets the minimum volume for a given source via the AL_MIN_GAIN source property.
 * @param al_source The source to set the volume to or 0 to set on all sources.
 * @param volume The new volume to use. Valid values are 0.0 to 1.0.
 * @return AL_TRUE on success or AL_FALSE on error.
 */
extern ALMIXER_DECLSPEC ALboolean ALMIXER_CALL ALmixer_SetMinVolumeSource(ALuint al_source, ALfloat volume);

/**
 * Gets the min volume via the AL_MIN_GAIN source property.
 * Gets the min volume for a given channel via the AL_MIN_GAIN source property.
 * @param which_channel The channel to get the volume from.
 * -1 will return the average volume set across all channels.
 * @return Returns the volume for the specified channel, or the average set volume for all channels, or -1.0 on error.
 */
extern ALMIXER_DECLSPEC ALfloat ALMIXER_CALL ALmixer_GetMinVolumeChannel(ALint which_channel);

/**
 * Gets the min volume via the AL_MIN_GAIN source property.
 * Gets the min volume for a given source via the AL_MIN_GAIN source property.
 * @param al_source The source to set the volume to or 0 to set on all sources.
 * 0 will return the average volume set across all channels.
 * @return Returns the volume for the specified channel, or the average set volume for all channels, or -1.0 on error.
 */
extern ALMIXER_DECLSPEC ALfloat ALMIXER_CALL ALmixer_GetMinVolumeSource(ALuint al_source);

/**
 * Sets the OpenAL listener AL_GAIN which can be thought of as the "master volume".
 * Sets the OpenAL listener AL_GAIN which can be thought of as the "master volume".
 * @param new_volume The new volume level to be set. Range is 0.0 to 1.0 where 1.0 is the max volume.
 * @return AL_TRUE on success or AL_FALSE on an error.
 */
extern ALMIXER_DECLSPEC ALboolean ALMIXER_CALL ALmixer_SetMasterVolume(ALfloat new_volume);

/**
 * Gets the OpenAL listener AL_GAIN which can be thought of as the "master volume".
 * Gets the OpenAL listener AL_GAIN which can be thought of as the "master volume".
 * @return The current volume level on the listener. -1.0 will be returned on an error.
 */
 extern ALMIXER_DECLSPEC ALfloat ALMIXER_CALL ALmixer_GetMasterVolume(void);

/**
 * @}
 */
 
/**
 * @defgroup QueryAPI Query APIs
 * @{
 * Functions to query ALmixer.
 */
 

/**
 * Returns true if the specified channel is currently playing or paused, 
 * or if -1 is passed the number of channels that are currently playing or paused.
 * @param which_channel The channel you want to know about or -1 to get the count of
 * currently playing/paused channels.
 * @return For a specific channel, 1 if the channel is playing or paused, 0 if not.
 * Or returns the count of currently playing/paused channels.
 * Or -1 on an error.
 */
extern ALMIXER_DECLSPEC ALint ALMIXER_CALL ALmixer_IsActiveChannel(ALint which_channel);

/**
 * Returns true if the specified source is currently playing or paused,
 * or if -1 is passed the number of sources that are currently playing or paused.
 * @param al_source The channel you want to know about or -1 to get the count of
 * currently playing/paused sources.
 * @return For a specific sources, 1 if the channel is playing or paused, 0 if not.
 * Or returns the count of currently playing/paused sources.
 * Or -1 on an error.
 */
extern ALMIXER_DECLSPEC ALint ALMIXER_CALL ALmixer_IsActiveSource(ALuint al_source);

/**
 * Returns true if the specified channel is currently playing. 
 * or if -1 is passed the number of channels that are currently playing.
 * @param which_channel The channel you want to know about or -1 to get the count of
 * currently playing channels.
 * @return For a specific channel, 1 if the channel is playing, 0 if not.
 * Or returns the count of currently playing channels.
 * Or -1 on an error.
 */
extern ALMIXER_DECLSPEC ALint ALMIXER_CALL ALmixer_IsPlayingChannel(ALint which_channel);

/**
 * Returns true if the specified sources is currently playing. 
 * or if -1 is passed the number of sources that are currently playing.
 * @param al_source The sources you want to know about or -1 to get the count of
 * currently playing sources.
 * @return For a specific source, 1 if the source is playing, 0 if not.
 * Or returns the count of currently playing sources.
 * Or -1 on an error.
 */
extern ALMIXER_DECLSPEC ALint ALMIXER_CALL ALmixer_IsPlayingSource(ALuint al_source);

/**
 * Returns true if the specified channel is currently paused. 
 * or if -1 is passed the number of channels that are currently paused.
 * @param which_channel The channel you want to know about or -1 to get the count of
 * currently paused channels.
 * @return For a specific channel, 1 if the channel is paused, 0 if not.
 * Or returns the count of currently paused channels.
 * Or -1 on an error.
 */
extern ALMIXER_DECLSPEC ALint ALMIXER_CALL ALmixer_IsPausedChannel(ALint which_channel);

/**
 * Returns true if the specified sources is currently paused. 
 * or if -1 is passed the number of sources that are currently paused.
 * @param al_source The source you want to know about or -1 to get the count of
 * currently paused sources.
 * @return For a specific source, 1 if the source is paused, 0 if not.
 * Or returns the count of currently paused sources.
 * Or -1 on an error.
 */
extern ALMIXER_DECLSPEC ALint ALMIXER_CALL ALmixer_IsPausedSource(ALuint al_source);

/**
 * Returns the number of channels that are currently available for playback (not playing, not paused).
 * @return The number of channels that are currently free.
 */
extern ALMIXER_DECLSPEC ALuint ALMIXER_CALL ALmixer_CountAllFreeChannels(void);

/**
 * Returns the number of channels that are currently available for playback (not playing, not paused),
 * excluding the channels that have been reserved.
 * @return The number of channels that are currently in free, excluding the channels that have been reserved.
 * @see ALmixer_ReserveChannels
 */
extern ALMIXER_DECLSPEC ALuint ALMIXER_CALL ALmixer_CountUnreservedFreeChannels(void);

/**
 * Returns the number of channels that are currently in use (playing/paused).
 * @return The number of channels that are currently in use.
 * @see ALmixer_ReserveChannels
 */
extern ALMIXER_DECLSPEC ALuint ALMIXER_CALL ALmixer_CountAllUsedChannels(void);

/**
 * Returns the number of channels that are currently in use (playing/paused),
 * excluding the channels that have been reserved.
 * @return The number of channels that are currently in use excluding the channels that have been reserved.
 * @see ALmixer_ReserveChannels
 */
extern ALMIXER_DECLSPEC ALuint ALMIXER_CALL ALmixer_CountUnreservedUsedChannels(void);


#ifdef DOXYGEN_ONLY
/**
 * Returns the number of allocated channels.
 * This is just a convenience alias to ALmixer_AllocateChannels(-1).
 * @see ALmixer_AllocateChannels
 */ 
ALuint ALmixer_CountTotalChannels(void);
#else
#define ALmixer_CountTotalChannels() ALmixer_AllocateChannels(-1)
#endif




#ifdef DOXYGEN_ONLY
/**
 * Returns the number of reserved channels.
 * This is just a convenience alias to ALmixer_ReserveChannels(-1).
 * @see ALmixer_ReserveChannels
 */ 
ALuint ALmixer_CountReservedChannels(void);
#else
#define ALmixer_CountReservedChannels() ALmixer_ReserveChannels(-1)
#endif


/**
 * @}
 */

/**
 * @defgroup DebugAPI Debug APIs
 * @{
 * Functions for debugging purposes. These may be removed in future versions.
 */
 

/* For testing */
#if 0
extern ALMIXER_DECLSPEC void ALMIXER_CALL ALmixer_OutputAttributes(void);
#endif
/** This function may be removed in the future. For debugging. Prints to stderr. Lists the decoders available. */ 
extern ALMIXER_DECLSPEC void ALMIXER_CALL ALmixer_OutputDecoders(void);
/** This function may be removed in the future. For debugging. Prints to stderr. */ 
extern ALMIXER_DECLSPEC void ALMIXER_CALL ALmixer_OutputOpenALInfo(void);

/** This function may be removed in the future. Returns true if compiled with threads, false if not. */ 
extern ALMIXER_DECLSPEC ALboolean ALMIXER_CALL ALmixer_CompiledWithThreadBackend(void);

/**
 * @}
 */




/* Ends C function definitions when using C++ */
#ifdef __cplusplus
}
#endif


#endif /* _SDL_ALMIXER_H_ */

/* end of SDL_ALmixer.h ... */